Device for acquiring and processing audible input

ABSTRACT

Embodiments of the disclosure generally include a method and apparatus for receiving and separating unwanted external noise from an audible input received from an audible source using an audible signal processing system that contains a plurality of audible signal sensing devices that are arranged and configured to detect an audible signal that is received from any position or angle within three dimensional space. The audible signal processing system is configured to analyze the received audible signals using a first signal processing technique that is able to separate unwanted low frequency range noise from the received audible signal and a second signal processing technique that is able to separate unwanted higher frequency range noise from the received audible signal. The audible signal processing system can then combine the signals processed by the first and second signal processing techniques to form a desired audible signal that has a high signal-to-noise ratio throughout the full speech range.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation-in-part of U.S. patent applicationSer. No. 15/650,614, filed Jul. 14, 2017, which claims the benefit ofU.S. provisional patent application Ser. No. 62/456,632, filed Feb. 8,2017, which are both herein incorporated by reference.

BACKGROUND OF THE INVENTION Field of the Invention

Embodiments of the present disclosure relate to a method and apparatusfor processing an audible signal to form a processed audible signal thathas an improved signal-to-noise ratio.

Description of the Related Art

The popularity and reliance on electronic devices has increaseddramatically in the past decade. The popularity of electronic devices,such as smart phones, touch pads, PDAs, portable computers, and portablemusic players, has increased in the past decade. Videotelephony andvideo conferencing devices have also become more popular in recentyears, thanks in large part to proliferation of high speed Internet andprice reductions in the supporting equipment. As the number ofelectronic devices and the reliance on these electronic devices hasincreased, there has been a desire for these devices to receive andprocess an audible input signal received from a user so that the audibleinput can be used to enable some desired task to be performed.

For years there has been a desire to construct machines that canrecognize, process and/or transmit various types of audible inputsreceived from a human being. Although in recent years this goal hasbegun to be realized, currently available systems have not been able toproduce results that are able to accurately detect these receivedaudible inputs in environments where external noise is common or notwell controlled. In most conventional microphone containing devices thatare configured to recognize and/or process various types of audibleinputs, it is often hard for the audible input processing electronics(e.g., voice recognition hardware) to clearly separate the desired humanspeech from the unwanted noise. This inability to separate audibleinputs from the surrounding noise within the environment is primarilydue to difficulties that are involved in extracting and identifying theindividual sounds that make up the human speech. These difficulties areexacerbated in noisy environments. Simplistically, speech may beconsidered as a sequence of sounds taken from basic sounds called“phonemes,” produced by a human. One or more phonemes represent a wordor a phrase. Thus, extraction of the particular phonemes containedwithin the received speech is necessary to achieve voice recognition,which is often extremely difficult in noisy environments.

Moreover, conventional voice or speech recognition hardware aretypically limited to detecting speech within the lower end of the speechfrequency range, such as between about 100 hertz (Hz) and about 3,000Hz, due to limitations in the device's sampling frequency and thegeometry of the microphone assemblies. Thus, a large amount of usefuldata is lost by these conventional designs since they are not able todetect speech throughout the full speech range which extends between 100Hz and about 8,000 Hz, and thus lose the information found in the higherend of the speech range found between 3,000 Hz and 8,000 Hz.

As the popularity of voice recognition systems increases, many usersutilize them in a variety of environments. Use of these various devicesis common in a myriad of moderately noisy to excessively noisyenvironments such as an office, conference room, airport, orrestaurants. Several conventional methods for performing noise reductionalready exist, however, many conventional methods can be categorized astypes of filtering. In the related art, speech and noise are acquired inthe same input channel, where they reside in the same frequency band andmay have similar correlation properties. Consequently, filtering willinevitably have an effect on both the speech signal and the backgroundnoise signal. Distinguishing between voice and background noise signalsis a challenging task. Speech components, which are received byconventional electronic devices, may be perceived as noise componentsand may be suppressed or filtered along with the noise components. Whilevoice recognition technology is increasingly sophisticated, a clearseparation of the voice component of an audio signal from noisecomponents, or in other words having a high signal-to-noise ratio (SNR),is required for acceptable levels of accuracy in the voice recognitionor even, in some cases, the delivery and reproduction of the receivedaudio signal at a distant location.

Additionally, as the number of electronic devices and the reliance onthese electronic devices has increased, there has been a desire forelectronic devices that are untethered to conventional wall outlet typesof power sources, thus allowing these untethered electronic devices tobe portable. However, the power supply in portable electronic devices iscommonly limited by a finite energy storage capacity provided by abattery. The rate of energy consumption by the device determines thetime of operation of the device until the battery needs to be rechargedor replaced. Therefore, it is desirable to find ways to reduce the powerconsumption used by the portable device's electronic components, such asvoice recognition elements, to improve the battery lifetime of theportable electronic devices.

Therefore, there is a need for an electronic device that solves theproblems described above. Moreover, there is a need for a portableelectronic device that is able to efficiently filter out unwanted noisefrom an audible input that is received from an audible source.

SUMMARY OF THE INVENTION

Embodiments of the disclosure generally include a method and apparatusfor receiving and separating unwanted external noise from an audibleinput received from an audible source. Embodiments of the disclosure mayinclude an audible signal processing system that contains a plurality ofaudible signal sensing devices (e.g., microphones) that are arranged andconfigured to detect an audible signal that is generated and provided tothe audible signal processing system from any position or angle withinthree dimensional (3-D) space.

BRIEF DESCRIPTION OF THE DRAWINGS

So that the manner in which the above recited features of the inventioncan be understood in detail, a more particular description of theinvention, briefly summarized above, may be had by reference toembodiments, some of which are illustrated in the appended drawings. Itis to be noted, however, that the appended drawings illustrate onlytypical embodiments of this invention and are therefore not to beconsidered limiting of its scope, for the invention may admit to otherequally effective embodiments.

FIG. 1A is a schematic diagram illustrating an example of an audiblesignal processing system, according to one or more embodiments of thepresent disclosure.

FIG. 1B is a plan view of the audible signal processing system shown inFIG. 1A without a center microphone shown, according to one or moreembodiments of the present disclosure.

FIG. 1C is a schematic block diagram of device components found withinthe audible signal processing system shown in FIG. 1A, according to oneor more embodiments of the present disclosure.

FIG. 2A is a diagram illustrating a plan view of an audible signalprocessing system and an audible source, according to one or moreembodiments of the present disclosure.

FIG. 2B is a graph depicting the time delay that each of the microphonesillustrated in FIG. 2A will experience when an audible signal isdelivered from the audible source to each of the microphones.

FIG. 3A is a graph depicting a composite audible signal that may bereceived by a microphone within the audible signal processing system,according to one or more embodiments of the present disclosure.

FIG. 3B includes graphs that depict the various components found withinthe composite audible signal illustrated in FIG. 3A.

FIG. 4A is a schematic diagram illustrating an example of an audiblesignal processing device found within the audible signal processingsystem, according to one or more embodiments of the present disclosure.

FIG. 4B is a schematic diagram illustrating portions of the audiblesignal processing device illustrated in FIG. 4A, according to one ormore embodiments of the present disclosure.

FIG. 4C is a schematic diagram illustrating a direction detectionelement of the audible signal processing device, according to one ormore embodiments of the present disclosure.

FIG. 4D illustrates a method of performing a first type of signalprocessing technique, according to one embodiment of the presentdisclosure.

FIG. 4E illustrates a method of determining a desired direction fromwhich an audible signal is received, according to one embodiment of thepresent disclosure.

FIG. 4F illustrates a method of performing a second type of signalprocessing technique, according to one embodiment of the presentdisclosure.

FIG. 5A is a plan view of an audible signal processing system havingthree microphones, according to an embodiment of the disclosure providedherein.

FIG. 5B is a plan view of the audible signal processing systemillustrated in FIG. 5A that further includes a plurality of directiondetection bins that may be used to determine the direction of an audiblesource, according to an embodiment of the disclosure provided herein.

FIG. 5C is a graph that illustrates weighting coefficients that can beused to determine the direction of an audible source, according to oneembodiment of the present disclosure.

FIGS. 6A-6C illustrate examples of cardioid patterns that are formed atdifferent frequencies by use of a first signal processing technique,according to an embodiment of the disclosure provided herein.

FIG. 6D is a plan view of an audible signal processing system havingseven microphones, according to an embodiment of the disclosure providedherein.

FIGS. 7A-7C illustrate examples of beamforming patterns that are formedat different frequencies by use of a second signal processing technique,according to an embodiment of the disclosure provided herein.

FIGS. 8A-8C illustrate examples of patterns that are formed at differentfrequencies by use of a first signal processing technique and a secondsignal processing technique, according to an embodiment of thedisclosure provided herein.

To facilitate understanding, identical reference numerals have beenused, where possible, to designate identical elements that are common tothe figures. It is contemplated that elements disclosed in oneembodiment may be beneficially utilized on other embodiments withoutspecific recitation. The drawings referred to here should not beunderstood as being drawn to scale unless specifically noted. Also, thedrawings are often simplified and details or components omitted forclarity of presentation and explanation. The drawings and discussionserve to explain principles discussed below, where like designationsdenote like elements.

DETAILED DESCRIPTION

In the following description, numerous specific details are set forth toprovide a more thorough understanding of the embodiments of the presentdisclosure. However, it will be apparent to one of skill in the art thatone or more of the embodiments of the present disclosure may bepracticed without one or more of these specific details. In otherinstances, well-known features have not been described in order to avoidobscuring one or more of the embodiments of the present disclosure.

Embodiments of the disclosure generally include a method and apparatusfor receiving and separating unwanted external noise from an audibleinput received from an audible source. Embodiments of the disclosure mayinclude an audible signal processing system that contains a plurality ofaudible signal sensing devices (e.g., microphones) that are arranged andconfigured to detect an audible signal that is generated and provided tothe audible signal processing system from any position or angle withinthree dimensional (3-D) space. The audible signal processing system isconfigured to analyze the audible signals received by each of theplurality of audible signal sensing devices using a first signalprocessing technique that is able to separate unwanted low frequencyrange noise from the detected audible signals and a second signalprocessing technique that is able to separate unwanted higher frequencyrange noise from the detected audible signals. The audible signalprocessing system is then configured to combine the signals processed bythe first and second signal processing techniques to form a desiredaudible signal that has a high signal-to-noise ratio throughout adesired frequency range, such as the full speech range.

In some configurations, the audible signal processing system is designedto be portable and thus run on a power source that has a finite amountenergy stored therein (e.g., battery). Therefore, in some cases theaudible signal processing system may be further configured to receiveand separate the unwanted external noise from a received audible inputin an efficient manner to extend the operation time of the portableaudible signal processing system. The audible signal processing systemmay also be configured to receive an audible signal from an externalsource, efficiently remove or separate unwanted noise from the receivedaudible signal, and then deliver the processed audible signal to asoftware application that is configured to further process and/orperform some desired activity based on the processed audible signal. Theaudible signal processing system may also be configured to deliver theprocessed audible signal to another electronic device that is configuredto receive and process the received information so that the seconddevice can perform some desired activity.

FIG. 1A is a schematic diagram illustrating an example of an audiblesignal processing system 100, according to one or more embodiments ofthe present disclosure. FIG. 1B is a diagram illustrating a plan view ofthe audible signal processing system 100 shown in FIG. 1A. The audiblesignal processing system 100 will generally include an electronic device102. The electronic device 102 can be a computing device that can beused as a standalone electronic device or an electronic device that canbe used in combination with other electronic devices. In one example, asshown in FIG. 1A, the electronic device 102 is able to communicate witha separate second electronic device 195 over a wired or wirelesscommunication link 194. Alternately, in another example, the electronicdevice 102 is a component within the second electronic device 195. Ineither case, the electronic device 102 and/or the second electronicdevice 195 may be a wireless speaker, video camera device that includesa CCD camera, a keyboard, smart phone, a speaker phone, home automationdevice, or other useful electronic device. In one example, theelectronic device 102 or the second electronic device 195 may be anUltimate Ears Boom™ speaker, a Harmony™ universal remote control or aLogitech Connect™ or a Logitech BCC 950™ video conferencing device thatare all available from Logitech USA of Newark Calif. or Logitech EuropeS.A of Lausanne, Switzerland. The electronic device 102 or the secondelectronic device 195 may also be an iPod®, iPhone®, iPad®, Android™phone, Samsung Galaxy®, Squeeze™box, Microsoft Surface®, laptop or othersimilar device. While the discussion below primarily describes orprovides examples of an audible signal processing system 100 that is astandalone portable electronic device this configuration is not intendedto be limiting as to the scope of the disclosure provided herein.

The electronic device 102 will include a plurality of audible signaldetection devices that are positioned in a geometrical array across oneor more surfaces of the electronic device 102. In some embodiments, thegeometrical array of audible signal detection devices, or hereafterreferred to as microphones 101, can be positioned in a two dimensional(2-D) array of microphones 101 or a three dimensional (3-D) array ofmicrophones, which may include microphones 101 and one or moremicrophones 121. The electronic device 102 may be any desirable shape,such as the cylindrical shape shown in FIG. 1A, and may include one ormore exterior surfaces on which one or more of the microphones 101, 121may be positioned, such as a top surface 106, a side surface 108 and asupporting surface 107. The supporting surface 107 is a surface on whichthe whole electronic device 102 may be positioned during normaloperation. Also, while FIGS. 1A-1B, 2A, 5A-5B, and 6D illustrate themicrophones 101, 121 being positioned on or extending from a surface ofthe electronic device 102, this configuration is not intended to belimiting as to the scope of the disclosure provided herein since themicrophones 101, 121 could also be positioned so that the exteriorsurface of the microphones are flush with or recessed from the surfaceon which they are disposed. While, for simplicity of discussion reasons,the disclosure below primarily includes a discussion relating to ageometrical array that includes a two dimensional (2-D) array ofmicrophones this configuration is not intended to be limiting as to thescope of the disclosure provided herein since three dimensional (3-D)microphone arrays can also be used with one or more of the embodimentsdisclosed herein. A geometrical array of microphones may include atleast three microphones, or, for example, five microphones 101, as shownin FIG. 1B, or even seven microphones as shown in FIG. 6D. However, insome cases it may be desirable to have only two microphones 101. Themicrophones 101 can be any type of electrical device that is able toconvert air pressure variations of a sound wave into an electricalsignal, and thus may include, but are not limited to a dynamicmicrophone, condenser microphone, piezoelectric microphone, fiber opticmicrophone, ribbon microphone, MEMS microphone or other similar device.In some embodiments, the microphones 101 are omnidirectional microphonesthat are able to detect audible signals from all directions.

In some embodiments, the microphones 101 are positioned in atwo-dimensional (2-D) geometrical array across the top surface 106and/or side surface 108 of the electronic device 102. In one example, asshown in FIGS. 1A and 1B, the geometrical array of microphones 101 areevenly distributed across the side surface 108. As illustrated in FIG.1B, five microphones 101 within the geometrical array are evenlydistributed in a circular array across the side surface 108 so that eachmicrophone 101 is positioned a distance 105 from a center point 103 andare at an equal angular distance 104 apart (e.g., 72° apart). Ingeneral, the microphones 101 within the electronic device 102 arepositioned so that all of the microphones 101 are positioned in ageometrical array (e.g., 2-D array or 3-D array) so that the electronicdevice 102 can better detect audible signals arriving from any directionand prevent aliasing of the detected direction of the received audiblesignal data that is common in conventional linear microphoneconfigurations. Thus, the electronic device 102 is able to receive anaudible signal “A” provided from an audible source 150 that ispositioned a distance in three dimensional space from the audible signalprocessing system 100 and then process the received audible signals sothat other unwanted audible signals that were provided from othersources that are positioned at different positions relative to theaudible signal processing system 100 can be preferentially excluded.

FIG. 1C is a schematic diagram illustrating an electronic assembly 135within the audible signal processing system 100, according to oneembodiment of the present disclosure. In general, the electronicassembly 135 will include a processor 118, non-volatile memory 122,power source 130 and an audible signal processing device 400. Duringoperation, the electronic device 102 is configured to receive aplurality of audible signals, which include an audible signal “A”, froma plurality of microphones 101. The audible signal processing device 400and processor 118 then receive the detected audible signals from each ofthe microphones 101 and processes the detected inputs to remove orseparate the unwanted external noise from the desired audible signal.The processed audible signal can then be used to perform some additionaltask by the audible signal processing system 100 or other downstreamdevice.

The audible signal processing device 400 generally includes electricalcomponents that can efficiently separate a desired portion on an audiblesignal from other received noise using a low frequency signal processingtechnique and a higher frequency signal processing technique. It isbelieved that the processes performed by the audible signal processingdevice 400 will reduce the error rate encountered when using theprocessed audible signal in a subsequent voice detection, voicecommunication, voice activated electronic device control and/or voicerecognition process versus processed audible signals generated byconventional noise cancelling or noise reduction techniques that arecommon today. The processes described herein are also adapted to extendthe operation time of the audible signal processing system 100 before arecharge or replacement of the power source 130 is required. While thepower source 130 described herein may include a battery, the electronicdevice 102 may at one time or another receive power from a wiredconnection to a wall outlet, wireless charger or other similar deviceswithout deviating from the basic scope of the disclosure providedherein.

The electronic assembly 135 may include the processor 118 that iscoupled to input/output (I/O) devices 116, the power source 130, and thenon-volatile memory unit 122. Memory unit 122 may include one or moresoftware applications 124, such as the controlling software programwhich is described further below. The memory unit 122 may also includestored media data 126 that is used by the processor 118 to performvarious parts of the methods described herein. The processor 118 may bea hardware unit or combination of hardware units capable of executingsoftware applications and processing data. In some configurations, theprocessor 118 includes a central processing unit (CPU), a digital signalprocessor (DSP), an application-specific integrated circuit (ASIC),and/or a combination of such units. The processor 118 is generallyconfigured to execute the one or more software applications 124 andprocess the stored media data 126, which may be each included within thememory unit 122.

The I/O devices 116 are coupled to memory unit 122 and processor 118,and may include devices capable of receiving input and/or devicescapable of providing output. The I/O devices 116 include the audioprocessing device 117 which receives the battery power and an inputsignal 104, and produces the output signal 106 which may be received andthen broadcast by the speaker system 111. The I/O devices 116 alsoinclude one or more wireless transceivers 120 that are configured toestablish one or more different types of wired or wireless communicationlinks with other transceivers residing within other computing devices. Agiven transceiver within the I/O devices 116 could establish, forexample, a Wi-Fi communication link, near field communication (NFC) linkor a Bluetooth® communication link (e.g., BTLE, Bluetooth classic),among other types of communication links with similar components in thesecond electronic device 195. In some embodiments, electronic componentswithin the I/O device 116 are adapted to transmit signals processed bythe audible signal processing device 400 to other internal electroniccomponents found within the audible signal processing system 100 and/orto electronic devices that are external to the audible signal processingsystem 100, as is discussed further below.

The memory unit 122 may be any technically feasible type of hardwareunit configured to store data. For example, the memory unit 122 could bea hard disk, a random access memory (RAM) module, a flash memory unit,or a combination of different hardware units configured to store data.The software application 124, which is stored within the memory unit122, includes program code that may be executed by processor 118 inorder to perform various functionalities associated with the electronicdevice 102. The stored media data 126 may include any type ofinformation that relates to a desired control parameter, quasi-directioninformation, calculated time delay information, noise signal RMSinformation, user data, electronic device configuration data, devicecontrol rules or other useful information, which are discussed furtherbelow. The stored media data 126 may include information that isdelivered to and/or received from the source 150 or another electronicdevice, such as the second electronic device 195. The stored media data126 may reflect various data files, settings and/or parametersassociated with the environment, audible signal processing devicecontrol and/or desired behavior of the electronic device 102.

As discussed above, during operation the electronic device 102 isconfigured to detect an audible signal “A” (e.g., voice command,acoustic signal) by use of a plurality of microphones 101 and thenprocess received audible signals using the audible signal processingdevice 400 so that the processed audible signals can be used to performsome desired task, or audible signal processing activity, by the audiblesignal processing system 100 or other electronic device, such as voicerecognition, voice communication, voice activated electronic devicecontrol and/or other useful audible signal enabled task or activity.However, depending on the position of the audible source 150 relative tothe microphones 101 within the electronic device 102 there will be adelay in the time when each microphone receives the same audible signal.In general, voice communication techniques will include any type oftwo-way communication process such as an audio chat, video chat, voicecall or other similar communication technique.

FIG. 2A illustrates one configuration in which the audible source 150 ispositioned a first distance 201A from a first microphone 101A, a seconddistance 201B from a second microphone 101B and a third distance 201Cfrom a third microphone 101C. Based on a far-field sound wavepropagation assumption the time delay seen by microphone 101B andmicrophone 101C relative to the first microphone 101A, which is closestto the audible source 150, will be equal to the distance 202A betweenthe first microphone 101A and the second microphone 101B in thedirection of the received audible signal and the distance 202B betweenthe first microphone 101A and the third microphone 101C in the directionof the received audible signal, respectively. FIG. 2B illustrates thedelays that will be seen by the microphones 101A-101C when they detectthe same audible signals 210A-210C, respectively, that are generated bythe source 150. However, the audible signals that are received by themicrophones 101A-101C will also receive audible signals from otherunwanted sources 155 at various different times due to each microphone'srelative position to the various unwanted sources 155. The signals fromthese unwanted sources 155 can prevent or obscure the electronic device102 from detecting the desired information found with the audible signalreceived from the source 150.

One will note that the delay one microphone will experience versusanother microphone is equal to the differences in distance of eachmicrophone from the source and the speed of sound (e.g., 340.3 m/s atsea level). As illustrated in FIG. 2B, the audible signal 210A isreceived by microphone 101A at time t_(A), and thus the delay that themicrophone 101B has when it receives the audible signal 210B from thetime when microphone 101A receives the audible signal 210A is equal tot_(B)−t_(A). The delay that the microphone 101C has relative tomicrophone 101A due to the time when it receives the audible signal 210Cversus when microphone 101A receives the audible signal 210A is equal tot_(C)−t_(A). Thus, the time delay that each microphone may see relativeto the other microphones within the geometrical array of microphoneswill depend on the relative orientation and position of the audiblesource to each of the microphones and their relative distance apart fromeach other. During the processing of the received audible signals by theaudible signal processing system 100, some additional signal processingrelated temporal delays, such as sampling rate delays, may be generated.

FIG. 3A illustrates a composite audible signal 301 that may be receivedby a single microphone within the array of microphones found within theelectronic device 102. The composite audible signal 301 will typicallycontain a desired audible signal provided from a desired audible sourceand a plurality of other audible signals received from other unwantedsources. FIG. 3B illustrates one possible configuration of a compositeaudible signal 301 that includes a constant audible noise signal 311that is provided from a first noise source (e.g., unwanted sources 155),a desired audible signal 312 that is provided from a desired audiblesource (e.g., source 150), and a second audible noise signal 313 that isprovided from a second noise source (e.g., unwanted sources 155). Thus,the composite audible signal 301 includes a plurality of audible signalsgenerated from a plurality of audible sources, which include the desiredaudible signal 312 from a desired audible source. In one example, asillustrated in FIG. 3A, the desired audible signal 312 is received by amicrophone between times t₁ and t₅ and has a varying intensity atdifferent frequencies over time. Separately, the constant audible noisesignal 311 may come from various common noise sources found inindustrial, office or even conference room environments, such as fans,lighting, HVAC units or other common audible sources, and may in somecases include a low frequency audible signal. The second audible noisesignal 313 may include audible signals that are generated by otheraudible sources, such as voices from other people in a room, musicplaying on a local speaker, or other unwanted noise sources. The secondaudible noise signal 313 will typically include audible signals providedat any frequency at any instant in time. In one example, as illustratedin FIG. 3B, the second audible noise signal 313 may extend between timest₂ and t₄ and have a varying frequency and intensity at different times.

One will note that the timing when each of the components of thecomposite audible signal 301 reach each microphone will differ in atleast one characteristic depending on the distance of the varioussources relative to each of the microphones within the array ofmicrophones found in the electronic assembly 102. In other words, forexample, the time when the second audible noise signal 313 and desiredaudible signal 312 overlap in time as detected by each microphone willdiffer, and thus the phase relationship and delay between each type ofreceived audible signal component will vary relative to each other frommicrophone to microphone.

Therefore, one goal of the audible signal processing device 400 withinthe electronic device 102 is to remove as much of the audible signalreceived from the first and second types of noise sources so that thedesired audible signal 312 can be separated therefrom. Once separated,the desired audible signal 312 can then be delivered to a softwareapplication that is configured to further process the desired audiblesignal so that some desired activity can be performed based on thereceipt of the desired audible signal 312. In some embodiments, thedesired audible signal 312 includes a user's speech that includesinformation across the full speech range, which typically extendsbetween about 100 Hz and about 8,000 Hz.

The design and configuration of the microphones within a geometric arraywithin the electronic device 102 can be made based on a balance of theneed to have a microphone array configuration that has a desired spacingto assure that the direction of the received audible signal can beaccurately determined, as will be discussed further below, versus theneed to assure that the signal processing technique (e.g., cardioidand/or beam forming) can preferentially reject unwanted noise across thefull speech range without the signal processing technique falling apartat either the higher end or the lower end of the frequency range. It isbelieved that most conventional spatial noise reduction techniques usedtoday are unable to work at or are ineffective at the high endfrequencies due to microphone spacing limitations or constraints, andthus most voice recognition or other similar programs are unable toeffectively utilize the information found in the higher end of thespeech range, such as between 4,000 Hz and 8,000 Hz.

FIGS. 4A-4C are schematic diagrams illustrating the various system levelcomponents that form the audible signal processing device 400 which isadapted to process the audible signals received from the audible signalprocessing system 100 illustrated in FIG. 5A. While the audible signalprocessing system 100 illustrated in FIG. 5A and most of the subsequentdiscussion below describes a configuration in which three microphones101A-101C are arranged in a planar circular array along the outersurface 108 of the audible signal processing system 100, thisconfiguration is not intended to be limiting as to the scope of thedisclosure provided herein. Other positions, orientations and numbers ofmicrophones could also be used to perform one or more aspects of thedisclosure provided herein.

FIG. 4A is schematic representation of one embodiment of the overallsystem that may be used to form the audible signal processing device400. The audible signal processing device 400 will include an optionalmicrophone gain element 420, a direction detection element 430, a firstsignal processor 405, a second signal processor 407, one or more postprocessing elements 451, 452 and a signal combining element 414. Thesignal combining element 414 will then provide the processed audiblesignal (e.g., desired audible signal) to a downstream element 415. Thedownstream element 415 may include a software application or otherelectronic device that uses the processed audible signal to perform somedesired activity. The various elements and/or components describedherein in conjunction with the audible signal processing device 400 maybe implemented by use of various analog and digital electricalcomponents that are used in combination with a controlling softwareprogram, or controlling software programs, that are executed by use ofthe processor 118, I/O devices 116 and memory unit 122. In someembodiments, the controlling software program(s) and various componentswithin the audible signal processing device 400, such as the directiondetection element 430, the first signal processor 405 and the secondsignal processor 407, can be brought into an active state by use of aphysical or audible command received from a user. In one embodiment, thecontrolling software program(s) and various components within theaudible signal processing device 400 are in a power saving idle stateuntil a physical or audible command is received from a user. In oneembodiment, the controlling software program(s) and various componentswithin the audible signal processing device 400 are in a power savingidle state until an audible command having a desired audible signallevel is received from a user. Referring to FIG. 3A, in one example, thecontrolling software program(s) compares a received composite audiblesignal 301 with a set audible signal level 321, which is stored as aparameter in memory, to determine if the received composite audiblesignal 301 exceeds the audible signal level 321, and thus a desiredaudible signal has been received from a user at one or more instants intime. In other embodiments, the controlling software program(s) isalways running in the background while the electronic device 102 ispowered on.

FIG. 4B is schematic representation of the various circuit elementsfound within the first signal processor 405 and second signal processor407, which will be discussed further below. FIG. 4C is schematicrepresentation of the various circuit elements found within thedirection detection element 430 of the audible signal processing device400, which will also be discussed further below.

The optional microphone gain element 420 typically includes microphonesignal gain adjusting elements that are adapted to adjust the signallevel of input received from each of the microphones within thegeometrical array of microphones. As illustrated in FIG. 4A, microphonesignal gain adjusting element 401A-401C are configured to separatelyadjust the signal level of the input received from each of themicrophones 101A-101C, respectfully, so that the incoming signal levelsprovided from each microphone are similar when they are processed by thevarious elements in the audible signal processing device 400. Amicrophone signal gain adjusting element 401 will include analog anddigital electrical circuit components that are configured and adjustedto provide a desired gain for an upstream microphone 101.

Referring to FIGS. 4A and 5A, the audible signal processing device 400is configured to process the audible signals received by the pluralityof the microphones 101A-101C using a first signal processing techniquethat is performed by the first signal processor 405 and a second signalprocessing technique that is performed by the second signal processor407. The audible signal processing device 400 is then configured tocombine the signals processed by the first and second signal processors405 and 407, by use of the signal combining element 414, to form adesired audible signal that has a high signal-to-noise ratio throughouta desired frequency range, such as the full speech range.Signal-to-noise ratio may be defined and/or measured as the ratio ofdesired signal power to the noise power, which is often expressed indecibels.

First Signal Processor Example

In some embodiments, the first signal processor 405 is configured toseparate unwanted low frequency noise from the detected audible signalsreceived from two or more of the microphones within the geometricalmicrophone array, while the second signal processor 407 is generallyconfigured to separate unwanted higher frequency noise from the detectedaudible signals received from all of the microphones within thegeometrical array.

In general, the first signal processor 405 is adapted to remove orseparate noise found within the lower end of the audible signalfrequency range from received composite audible signal using a cardioidnoise rejection technique. In order to perform the first signalprocessing technique, the first signal processor 405 will include or useportions of the controlling software program and various analog anddigital hardware components to perform the desired processes describedherein. In some embodiments, the first signal processor 405 includeselements that are formed within a digital signal processor (DSP) module.In some embodiments, the first signal processor 405 is adapted to removeor separate the unwanted noise from a desired audible signal using acardioid signal processing technique. The cardioid signal processingtechnique performed by the first signal processor 405 is generallyadapted to reject noise received from an off axis direction relative tothe direction of a desired audible signal source using a pattern that issimilar to an endfire cardioid. FIGS. 6A-6C illustrate the variousshapes of cardioid based patterns generated using two microphones, whichare a fixed distance apart, at various different frequencies. As will bediscussed further below, the controlling software used within theaudible signal processing device 400 may utilize two or more microphoneswithin the geometrical array to separate unwanted low frequency noisefrom the detected audible signals using a first-order cardioid,second-order cardioid, hyper-cardioid, super-cardioid or other similarcardioid technique.

In some embodiments, a first-order cardioid is formed by the firstsignal processor 405 by use of two audible signal inputs that arepositioned along a direction that is in-line with the direction that theaudible signal is received from the audible signal source. In oneembodiment, the first-order cardioid is formed using two audible signalinputs that are received from two of the microphones found within thegeometrical array of microphones. Additionally or alternately, as willbe discussed further below, the cardioid pattern is formed by averagingthe inputs from two microphones to form a virtual microphone audiblesignal and then using the formed virtual microphone's audible signal andan audible signal from one of the other microphones in the geometricalarray to from the first-order cardioid. While other higher-ordercardioid forming signal processing techniques could be used by the firstsignal processor 405, it is believed that the use of a first-ordercardioid for the low frequency signal processing has advantages overthese higher-order cardioid signal processing techniques. In general,from a power conservation stand point, it is desirable to use a fewernumber of microphones to perform the signal processing techniquesdisclosed herein. Thus, a first-order cardioid generation technique hasadvantages over other signal processing techniques that need anincreased number of microphones to form the higher order cardioidpatterns. One will note that a large portion of the power consumption iscreated by the process of reading, comparing and then writing to abuffer the audible signals, and performing other related pointer math,for each of the microphones within the electronic device at the highsampling rates required to perform these types of signal processingtechniques. In another example, it is believed that higher-ordercardioid signal processing techniques will require additional computingpower and time to process the audible signals received from three ormore microphones. The additional amount of computing power can thuscreate a significant drain on a battery powered type of power source130.

In some embodiments, the first order cardioid noise rejection techniqueutilized by the first signal processor 405 can be achieved by summingaudible signals received by two microphone elements within thegeometrical array of microphones, where one of the audible signals isinverted and delayed a period of time relative to the other receivedaudible signal before the two audible signals are then summed together.The amount of time delay generated by the first signal processor 405 isrelated to the speed of sound and the effective distance between themicrophones in the direction that the desired audible signal isreceived. The first signal processing technique is able to form adesired cardioid pattern for rejecting unwanted noise received fromoff-axis orientations as long as the wavelength is much longer than thedistance between the two microphones used to form the cardioid pattern.However, the ability to reject unwanted noise in a cardioid patterndegrades once the wavelength approaches a proportional distance betweenthe microphones. Therefore, the closer the microphones are to eachother, the higher in frequency the cardioid pattern can be maintained.

FIG. 6A illustrates an example of a pattern 601 that is formed by thefirst signal processing technique using two microphones that are a fixeddistance apart (e.g., 70 mm) at a first frequency of about 100 Hz. Onewill note that a desired audible source would be desirably positioned atthe 90° position relative to the orientation of the polar graph, suchthat the maximum amount of audible signal is accepted from thisdirection while the amount of audible signal that is accepted from otheroff-axis directions decreases all the way to zero at the −90° direction(e.g., 180° from the desired audible source's direction). FIG. 6A alsoincludes a second pattern 602 that is formed at a second frequency ofabout 1200 Hz using the same two microphones. One will note that bychanging the frequency from 100 Hz to about 1200 Hz the ability of thefirst signal processing technique to reject off-axis noise sources, suchas noise sources positioned outside of the angular range extendingbetween about 45° and about 135°, decreases a noticeable amount as thedetected audible signal frequency has increased. Therefore, geometricalmicrophone arrays that contain the spacing used to form the patternsshown in FIGS. 6A-6C will be ineffective at frequencies that exceed thepoint where the cardioid pattern breaks down (e.g., ˜1200 Hz), and thus,in this example, the first signal processing technique will generallynot be effective for use with audible signals that contain frequenciesgreater than 1200 Hz.

FIG. 6B illustrates an example of a third pattern 603 that is formed bythe first signal processing technique at a third frequency of about 2000Hz using the same two microphones used to form the pattern 601. In thisexample, the third pattern 603 has become distorted due to the effect offrequency on the detected signal such that it has formed two lobes 603Aand 603B. The third pattern 603 is substantially different from thepattern 601 illustrated in FIG. 6A, since the ability of the firstsignal processing technique to reject off-axis noise sources in the 15°and 165° directions due to the lobes 603A and 603B will be non-existent,while the first signal processing technique will tend to reject nearlyhalf of the audible signal received from a desired audible source thatis positioned at the 90° position.

FIG. 6C illustrates an example of a fourth pattern 604 that is formed ata fourth frequency of about 8000 Hz by the same two microphones used toform the first pattern 601. In this example, fourth pattern 604 of thefirst order mic array has also become distorted due to the effect offrequency on the detected signal such that it contains seven lobes604A-604G. Therefore, the ability of the first signal processingtechnique to reject off-axis noise sources in the seven directions ofthe lobes 604A-604G (e.g., 0°, 45°, 135°, etc.) will be minimal, whilethe first signal processing technique will tend to reject less thanabout 5 dB of the audible signal received from a desired audible sourcethat is positioned at the 90° position.

While the discussion surrounding FIGS. 6A-6C all discuss the effect thatfrequency has on the ability of the first signal processing technique toreject off-axis noise sources, similar effects can be seen by adjustingthe hardware design such as the spacing between the two microphones usedto form the cardioid based pattern. For example, a similar pair ofpatterns 601 and 602 (FIG. 6A) can be formed when the distance betweenthe two microphones is reduced from a distance of 70 mm to about 35 mmusing the same measurement frequency of about 1100 Hz. Therefore, insome embodiments, the relative spacing between the various microphonesmay be adjusted in the audible signal processing system 100 so that theupper end of the frequency range that the first signal processingtechnique can operate is desirably set. In some embodiments, the upperend of the frequency range of the first signal processing technique isset so that the transition to the frequency range that the second signalprocessing technique can perform its desired function (i.e., removehigher frequency noise) at least partially overlap. The process ofadjusting the upper end of the first signal processing technique and thelower end of the second processing technique is discussed further belowin conjunction with FIGS. 8A-8C. Alternately, in some embodiments, theupper end of the frequency range of the first signal processingtechnique is set to a level where the generated cardioid based patterndoes not become significantly distorted, such as where two or more lobesare formed and/or the attenuation of the desired audible signal that isdelivered in a desired audible source's direction (e.g., 90° in FIG.6A-6C) is less than about 1 to 5 dB. For example, the upper end of thefrequency range may be selected such that the wavelength is on the orderof the distance between the microphones used to form the patterngenerated by the first signal processing technique. Alternately, inanother embodiment, the upper end of the frequency range is betweenabout 500 Hz and about 4,000 Hz, such as in a frequency range betweenabout 1000 Hz and about 4,000 Hz. In some examples, the upper end of thefrequency range is about 700 Hz, about 1000 Hz, about 1500 Hz, about2000 Hz, or even about 4000 Hz.

Referring back to FIG. 4B, the first signal processor 405 generallyincludes a microphone input selection element 440, a signal delayingelement 441, a signal inverting element 442, a signal combining element443 and a signal filtering element 411, a parametric Equalizer 412 and amix gain amplifier 413 within the post processing element 452. FIG. 4Dillustrates a method 470 of performing a first type of signal processingtechnique using the first signal processor 405. During the operation, atstep 472, the first signal processor 405 first receives detecteddirection information from the direction detection element 430 based onan analysis of audible signals received by the geometrical array ofmicrophones. The detected direction information then allows the cardioidgeneration process to be performed so that off-axis noise can berejected as described above. Referring to FIGS. 5A and 6A, in oneexample, if the detected direction of the desired audible source isfound to be in the 90° direction by an analysis performed by thedirection detection element 430, a cardioid pattern that is orientedsimilarly to the patterns 601 and 602 will be generated by the firstsignal processor 405.

Next at step 473, the microphone input selection element 440 selects twomicrophones within the geometrical array of microphones based on thedetected direction information received from the direction detectionelement 430. In this example, the first signal processor 405 performs ananalysis of the audible signals received by the microphones 101B and101C, since they are aligned along a line 505B that extends in the 90°and −90° directions.

After the desired pair of microphones has been selected, steps 474 and475 are completed, which includes the delivery of the audible signalreceived from the microphone furthest from the audible signal source(e.g., microphone 101B) to the signal delaying element 441 and a signalinverting element 442, by the delivery of the audible signal along path440B of the first signal processor 405, to form a delayed and invertedaudible signal. The time delay used by the signal delaying element 441is set by the known distance between the two microphones based on theknown speed of sound. The time delay value used may be stored andretrieved from the memory unit 122.

Next, at step 476, the “undelayed” audible signal received from themicrophone closest to the audible signal source (e.g., audible signalreceived from microphone 101C along path 440A) and the delayed andinverted audible signal are then combined together by use of the signalcombining element 443.

Next, at step 477, the combined signals are then optionally filtered byuse of the signal filtering element 411 within the post processingelement 452. The signal filtering element may include a low pass filterand or a high pass filter that are able to remove frequencies that arehigher and/or lower than the useable signal processing range of thefirst signal processor 405, such as frequencies where the cardioidpattern become significantly distorted. For example, a low pass filterfrequency may coincide with a frequency between about 1,000 Hz and about4,000 Hz, and a high pass filter frequency may coincide with a frequencyof about 100 Hz, or, for example, with a frequency of about 700 Hz, orabout 1,000 Hz, or even about 2000 Hz.

After the processes performed in method 470 have been completed thefirst signal processor 405 will then provide the processed audiblesignal, or hereafter first processed audible signal, to the signalcombining element 414 where, in step 498, the first processed audiblesignal and a second processed audible signal received from the secondsignal processing technique are combined. As will be discussed furtherbelow, at step 499, the combined first and second processed audiblesignals are then transferred to other downstream electronic devices orelements within the audible signal processing system 100.

Therefore, based on the geometrical array of microphones illustrated inFIG. 5A, audible signal sources that are aligned in the directionsparallel to line 505A, line 505B or line 505C can be accuratelyrepresented by a generated cardioid pattern. Line 505A can be used whenthe audible signal sources are aligned in directions −30° and 150°. Line505B can be used when the audible signal sources are aligned indirections 90° and −90°. Line 505C can be used when the audible signalsources are aligned in directions 30° and −150°. However, audiblesources that are not aligned with the directions illustrated by lines505A-505C will have less accurate noise rejection due to the off angleerror produced when the misaligned cardioid pattern is generated by thefirst signal processor 405.

In an effort to reduce the off-angle error produced when the selectedmicrophones are not aligned in the exact direction of the audiblesource, a virtual microphone can be used to minimize or effectivelyeliminate any error created by the misalignment of the formed cardioidpattern relative to the audible source. A virtual microphone can becreated by combining the audible signals received by two microphoneswithin the geometric array to form a microphone that is effectivelypositioned at a point along a line that extends between the twomicrophones. The combined audible signal will generally approximate aportion of an audible signal that would have been received by a virtualmicrophone that is positioned between the first and second microphones.The process of combining the audible signals may include averaging, oreven weighting, the two audible signals received by the two selectedmicrophones within the geometric array to form a virtual microphoneaudible signal that can then be used by the first signal processingtechnique in a similar way that an audible signal from an “actual”microphone is used. For example, referring to FIG. 5A, if the directiondetection element 430 determines that the audible source is positionedat a direction of about 60°, the first signal processor 405 can thenaverage the audible signals received by the first microphone 101A andthe third microphone 101C to form a virtual microphone 520 that ispositioned at the midpoint of the portion of the line 505A that extendsbetween the two microphones. The first signal processor 405 can then usethe generated virtual microphone 520 and a second microphone (e.g.,microphone 101B) to complete the first-order cardioid generationprocess. In this example, the first signal processor 405 performs ananalysis of the audible signals effectively received by the virtualmicrophone 520 and microphone 101B, since these microphones are alignedalong the line 522 that extends in the 60° and −120° directions. Afterthe desired pair of microphones has been selected (e.g., step 473), thecardioid generation process will then include the delivery of theaudible signal received from the microphone 101B to the signal delayingelement 441 (e.g., step 474) and a signal inverting element 442 (e.g.,step 475) by the delivery of the audible signal along path 440B to forma delayed and inverted audible signal. The time delay value used by thesignal delaying element 441 is set by the distance between these twomicrophones, which in this case is the distance 521, and the known speedof sound. The time delay value that is used may be stored and retrievedfrom the memory unit 122. One will note that the distance 521 is smallerthan the distances 510A-510C, and thus will have an effect on thefrequency limit for the first signal processing technique, and thus maybe accounted for by the first signal processor 405. Next, the“undelayed” audible signal received from the microphone closest to theaudible signal source (e.g., virtual microphone 520) and the delayed andinverted audible signal received from the second microphone 101B arethen combined together (e.g., step 476) by use of the signal combiningelement 443 and then filtered by use of the signal filtering element 411within the post processing element 452. After the processes withinmethod 470 have been performed, the first signal processor 405 will thenprovide the virtual microphone containing processed audible signal tothe signal combining element 414 where, in step 498, the first processedaudible signal and a second processed audible signal received from thesecond signal processing technique are then combined. As will bediscussed further below, at step 499, the combined first and secondprocessed audible signals are then transferred to other downstreamelectronic devices or elements within the audible signal processingsystem 100.

Therefore, by generating virtual microphones that are positioned along aline that extends between two microphones, the maximum off angle errorthat is produced when the selected microphones are not aligned in theexact direction of the audible source can be reduced by at least half ofthe angle formed between the microphones used to form the virtualmicrophone. For example, the virtual microphone in the above examplewould thus effectively have a microphone positioned every 60° versus theactual 120° distance between each of the three actual microphones101A-101C. Thus, by use of a microphone signal averaging technique toform the virtual microphone the first signal processing technique can beused to detect audible signals found in at least 12 different directionswhile using only 3 microphones. Alternately, for virtual microphonegeneration techniques that use a weighted sum of the audible signalsreceived by two or more microphones the first signal processingtechnique could find an infinite number of possible directions toposition the virtual microphone along the line extending between themicrophones while using only 3 microphones. The weighting values used tocreate the weighted sum of the audible signals may be based on acomparison of the determined direction received from the directiondetection element 430 and its relationship (e.g., relative angle) to oneof the six directions that are parallel to lines 505A, 505B or 505C.Therefore, as noted above, by use of the virtual microphone technique afewer number of microphones are needed to perform the first signalprocessing technique and thus less power will be consumed by theelectronic device in performance of this signal processing technique.

In some virtual microphone generating embodiments, it is desirable toselect two microphones that are positioned along a line that issubstantially perpendicular to the direction that an audible signal isreceived to form the virtual microphone. Also, by increasing the numberof actual microphones found within the geometrical array of microphonesthe need for the generation of virtual microphones can be reduced, sincethe error will be reduced.

Alternate First Signal Processing Example

Alternately, in electronic device configurations that are not limited byelectrical power constraints (e.g., electrical power is received from awall plug or large battery) and/or limited by the processor's speed andother processor related resources it may be desirable to remove orseparate noise found within the audible signal frequency range by use ofa cardioid noise rejection technique that uses a higher order cardioidthan a first order based cardioid signal processing technique. In someembodiments, the first signal processor 405 is adapted to remove noiseusing a second-order or greater cardioid based signal processingtechnique. In one configuration, the cardioid signal processingtechnique performed by the first signal processor 405 is adapted toreject noise received from an off-axis direction relative to the audiblesignal's direction using three or more microphones within thegeometrical array. The controlling software used within the audiblesignal processing device 400 may thus utilize three or more microphonesand/or generated virtual microphones that are aligned in a desireddirection to separate unwanted noise from the detected audible signalsusing a cardioid pattern.

In some embodiments, the first signal processing technique is adapted toform higher order cardioid patterns by use of three or more audiblesignal inputs that are positioned along a direction that is in-line withthe direction that the audible signal is received from the audiblesignal source. In one embodiment, the higher order cardioid is formedusing the three or more audible signal inputs that are received from themicrophones found within the geometrical array of microphones.Additionally or alternately, the higher order cardioid patterns areformed by combining (e.g., averaging or weighting) the inputs from twoor more microphones to form a virtual microphone audible signal and thenusing the formed virtual microphone's audible signal and an audiblesignal from one of the other microphones or other virtually formedmicrophones to from the higher order cardioid.

FIG. 6D illustrates an example of a geometrical array configuration thatmay be used by the first signal processing technique to form a desiredcardioid pattern. In this example, the audible signal processing system100 includes seven microphones 601A-601G that are arranged in ageometrical array, which contains six microphones disposed along theouter surface 108 (e.g., microphones 601A-601G) and at the center point103 of the top surface 106 (e.g., microphone 601G) of the electronicdevice 102. However, one skilled in the art will appreciate that othergeometrical array configurations can be used by the first signalprocessing technique to form higher order cardioid patterns. In oneexample, the geometrical array may include five microphones that arearranged in a configuration that includes four microphones disposedalong the outer surface 108 and one microphone at the center point 103.In some embodiments, the geometrical array includes a plurality ofmicrophones that are only disposed along the outer surface 108 of theelectronic device 102, such as an array that contains only sixmicrophones 601A-601G (FIG. 6D), or other useful odd or even number ofmicrophones in the geometrical array.

During operation, the first signal processor 405 receives audible signaldirection information from the direction detection element 430 notingthat audible signals are being received from a direction 605 (i.e., 0°direction). The first signal processor 405 will then determine by use ofthe controlling software and microphone input selection element 440 thatmicrophone 601G, a virtual microphone 620A, and a virtual microphone620B are needed to perform the first signal processing technique, sincethese microphones are aligned along the direction 605. In this example,once the controlling software has formed the virtual microphone 620A byaveraging the audible signals received by microphones 601A and 601F, andvirtual microphone 620B has been formed by averaging the audible signalsreceived by microphones 601C and 601D, the process of forming thedesired cardioid pattern can be completed. In this example, the virtualmicrophone 620A is positioned at the midpoint of line 621A and thevirtual microphone 620B is positioned at the midpoint of line 621B.

Next, the first signal processor 405 then uses a cascaded cardioidgeneration process to perform the first signal processing technique. Thecascaded cardioid generation process begins by the delivery of theaudible signal received from the microphone 601G to the signal delayingelement 441 (e.g., step 474) and the signal inverting element 442 (e.g.,step 475) to form a first delayed and inverted audible signal. The timedelay value that is used may be stored and retrieved from the memoryunit 122 for this known microphone configuration. The “undelayed”audible signal received from the virtual microphone 620A and the firstdelayed and inverted audible signal received from the second microphone601G are then combined together (e.g., step 476) by use of the signalcombining element 443 to form a first combined cascaded audible signalthat is stored within memory.

The cascaded cardioid generation process then continues on to form asecond combined cascaded audible signal by delivering the audible signalreceived from the second virtual microphone 620B to the signal delayingelement 441 (e.g., step 474) and the signal inverting element 442 (e.g.,step 475) to form a second delayed and inverted audible signal. The“undelayed” audible signal received from microphone 601G and the seconddelayed and inverted audible signal are then combined together (e.g.,step 476) by use of the signal combining element 443 to form a secondcombined cascaded audible signal that is stored within memory.

The cascaded cardioid generation process then delivers the secondcombined cascaded audible signal to the signal delaying element 441(e.g., step 474) and the signal inverting element 442 (e.g., step 475)to form a first combined delayed and inverted audible signal. The timedelay value that is used may be stored and retrieved from the memoryunit 122. Then, the first combined cascaded audible signal and the firstcombined delayed and inverted audible signal are then combined together(e.g., step 476) by use of the signal combining element 443 and thenfiltered by use of the signal filtering element 411 to form the completecombined cascaded audible signal.

After a complete combined cascaded audible signal has been formed, thefirst signal processor 405 will then provide the complete combinedcascaded audible signal to the signal combining element 414 where, instep 498, the complete combined cascaded audible signal and a secondprocessed audible signal received from the second signal processingtechnique are combined. As will be discussed further below, at step 499,the combined signals are then transferred to other downstream electronicdevices or elements within the audible signal processing system 100.

In cases where the audible signal source is aligned with a directionthat includes three “actual” microphones, such as direction 606 in FIG.6D, the cascaded cardioid generation process can be performed using onlythe three microphones, and thus no virtual microphones. In this example,the first signal processor 405 will then determine by use of thecontrolling software and microphone input selection element 440 thatmicrophone 601F, microphone 601G, and microphone 601C are needed toperform the first signal processing technique. Then, as similarlydiscussed above, the first signal processing technique then uses thethree microphones and above described process steps to similarly formthe complete combined cascaded audible signal.

In electronic device configurations that include an array of microphoneswhich are primarily disposed along the outer surface 108 of theelectronic device 102, such as configurations where no centrallypositioned microphone exists, a virtual microphone can be used in placeof a centrally positioned microphone by averaging the audible signalsreceived by two or more microphones within the geometric array.Referring to FIG. 6D, in one example, a virtual microphone can be formedat the center point 103 by averaging the audible signals received bymicrophones 601B and 601E. Thus, in some cases, a cascaded cardioidgeneration process can be performed using only two “actual” microphonesand one virtual microphone. Therefore, in relation to the previousexample, the first signal processor 405 can determine by use of thecontrolling software and microphone input selection element 440 thatmicrophone 601F, microphone 601C, microphone 601B and microphone 601Eare needed to perform the first signal processing technique. In thisexample, microphones 601B and 601E are used to form the centrallypositioned virtual microphone, and then the formed virtual microphoneand microphones 601F and 601C are used to perform the first signalprocessing technique. Then, as similarly discussed above, the firstsignal processing technique then uses these “actual” and virtualmicrophones and the above described process steps to similarly form thecomplete combined cascaded audible signal.

Direction Detection Process and Elements

Referring to FIGS. 4A and 4C, during the operation of the audible signalprocessing device 100 the direction detection element 430 is configuredto determine and provide detected direction information to the firstsignal processor 405 and the second signal processor 407. FIG. 4Eillustrates a method 480 of performing a direction detection techniqueusing the direction detection element 430. The direction detectionelement 430 generally includes or uses portions of the controllingsoftware program and various analog and digital hardware components toperform one or more of the processes described herein. In someembodiments, the direction detection element 430 includes elements thatare formed within a digital signal processor (DSP) module. In someembodiments, the direction detection element 430 includes an equalizingelement 460, an RMS processing element 467, a delay analysis element462, a direction determination element 463, a coefficient generationelement 464, a direction selection element 465, and a direction deliveryelement 466.

During operation, at step 482, after receiving the microphone inputsfrom the optional microphone gain element 420 during step 481, each ofthe audible signals provided from each of the microphones 101A-101C arepreprocessed. The preprocessing steps may include processing the audiblesignal using an equalizing element 460 so that a certain desiredfrequency range, which may be associated with the speech range, can beextracted or isolated from other unwanted frequency regions before it isprocessed by the subsequent direction detection elements. In someconfigurations, the equalizing element 460 may include parametricequalizers 461A-461C that are each configured to process the audiblesignal received from each of the microphones 101A-101C, respectively. Insome embodiments, the equalizing element 460 is configured topreferentially allow frequencies within the full speech range to passtherethrough and be delivered to one or more downstream components.

Next, at step 483, at least one of the inputs from a microphone withinthe geometrical array of microphones is then delivered to theroot-mean-square (RMS) processing element 467 that is used to detect andin some cases remove constant audible noise signals 311 found within thecomposite audible signals 301 that are received by each of themicrophones. The RMS processing element 467 may utilize an RMS thresholdanalysis element 468 that contains and/or monitors and storesinformation regarding the level of the constant audible noise signal 311over time. The RMS processing element 467 may be configured to detectthe current level of the received constant audible noise signal 311, andfurther receive input regarding historic constant audible noise signallevels from the RMS threshold analysis element 468, so that unwantedbackground noise does not get utilized in the direction detectionalgorithm and thus affect the direction detection element's results. Insome embodiments, the RMS threshold analysis element 468 uses themeasured level of the constant audible noise and compares it to areceived audible signal to determine if the incoming audible data shouldbe used towards the determination of its arrival direction, and thus thereceived audible data is not part of the noise within the environment110 (FIG. 1A). The RMS processing element 467 can thus be used to reducethe risk of background music or crowd noise from affecting the directionanalysis that is utilized by the first and second signal processingtechniques.

Next, at step 484, the delay analysis element 462 receives the audiblesignals processed by the equalizing element 460 and analyzes the audiblesignals to determine the relative delays in the receipt of the audiblesignal experienced by each of the microphones. In some embodiments, thedelay analysis element 462 analyzes each of the received audible signalsas a function of time to determine which attributes in each of thereceived audible signals are common in each of the received audiblesignals to determine the relative delay experienced by each microphone.In one configuration, as illustrated in FIG. 2B, the delay analysiselement 462 may detect a change in the sound level or signal pattern asa function of time to determine the delay experienced by each of themicrophones. In one example, the controlling software program(s) may usea cross-correlation technique to determine the delay in the receipt ofthe audible signal experienced by each microphone, even though thereceived audible signal may be part of a composite audible signal thatis received by the microphones. In another example, referring back toFIG. 3A, the controlling software program(s) may compare the receivedcomposite audible signals 301 from the microphones within geometricarray with the set audible signal level 321 to determine if at least aportion of the received composite audible signal 301 exceeds the audiblesignal level 321 for at least one of the microphones, and thus a desiredaudible signal that has been received from a user at one or moreinstants in time is singled out based on its higher signal level (e.g.,dB level), and in some cases additionally using a cross-correlationtechnique. In some cases, to reduce device power consumption and/orreduce the number of tasks performed by the processor, it may bedesirable to only utilize the cross-correlation technique when the soundlevel of the received composite audible signal 301 exceeds the audiblesignal level 321. One will note that the microphone that is closest tothe audible signal source will contain a zero time delay and the othermicrophones will contain a time delay that is related to the distance ofthe microphone from the closest microphone in the direction that theaudible signal is received from the audible source. In some cases, thecontrolling software program will assure that the delay times are allpositive and store the relative delays in the electronic device'smemory.

At step 485, after determining the relative delay of each of themicrophones, the direction determination element 463 then determines thedirection that the audible signal is being received from by use of oneor more portions of the controlling software program. While it ispossible to perform various complicated mathematical analysis techniquesto determine the exact position of the audible source relative to theelectronic device 102, it has been found that these highly analyticaldirection detection processes require a significant amount of computingpower and time, and thus can create a significant drain on the powersource 130. The incorporation of these types of highly analyticaldirection detection processes would greatly increase the cost andcomplexity of a consumer electronic device that is able to perform thesetypes of direction detection processes.

Therefore, in some embodiments, the direction determination element 463utilizes a less analytically intensive and power intensive statisticalbinning approach to determine the direction of the audible source. Whilethe direction determination element 463 in some cases will be able todetect the exact direction of the audible source from the electronicdevice 102, in most cases the direction that is determined by thedirection determination element 463 will have an error, which at itsmaximum is related to the sample rate of the analysis program, spacingof the microphones and also the size of the direction bins selected andused by the controlling software program(s) to determine the nearestdirection to the audible source direction. Thus, an audible signaldirection determined by the statistical analysis that is performed bythe direction determination element 463 is described herein as a“quasi-direction.”

During the direction determination process the direction determinationelement 463 uses the relative delay times received from the delayanalysis element 462 to determine the direction of the received audiblesignal. The controlling software program and the direction determinationelement 463 may break-up the pattern of the geometrical array ofmicrophones into binned regions. The number of binned regions willtypically relate to the number of microphones that are contained withinthe geometrical array of microphones as well as the minimum width of thebeam. In one example, the electronic device 102 illustrated in FIG. 1Bcould be divided up into five sectors or five angular regions. FIG. 5Bis a plan view of the audible signal processing system 100 illustratedin FIG. 5A that further includes a plurality of direction detectionregions that may be used to determine the direction of an audiblesource. In one example, the electronic device 102 can be broken up intothree sectors 531A, 531B and 531C, which each include an angular regionthat has a 120° angular distance as measured from a vertex 536. In oneconfiguration, the vertex 536 is positioned at the geometric center ofthe geometrical array of microphones. However, the vertex 536 may bepositioned in other desired geometric locations relative to thegeometric array of microphones to simplify the analysis performed by thecontrolling software program. Next, the direction determination element463 then determines which region of the geometrical array of microphonesthe audible source is positioned nearest to currently. In one example,if the audible source is positioned at an angle of about 35° relative tothe electronic device 102 illustrated in FIG. 5B, the directiondetermination element 463 will be able to first determine that theaudible source is positioned within sector 531A, which extends between0° and 120°, since the delay analysis element 462 will have determinedthat microphone 101A is the closest microphone to the audible source,and microphone 101C has a smaller time delay than microphone 101B due totheir relative positions to the audible source.

Having determined the region of the geometrical array of microphonesthat the audible source is positioned nearest, the directiondetermination element 463 will then determine which directional bin (orbinned region) within the determined region the audible source'sdirection is closest to so that a nearest quasi-direction can bedetermined. The directional bins are formed by dividing the angularregion or sector into a desired number of sub-regions that meet desiredaccuracy and computing power goals. For example, each of the sectors531A, 531B and 531C may be divided up into four binned regions that areeach separated by a 30° interval. In one example, the first sector531A's four bin configuration can be divided so that the edges of eachof the bins have a known quasi-direction, such as directions 0°, 30°,60°, 90° and 120° being the edges between the four formed bins. Thus,the angular distance formed for each defined bin (e.g., 30° bin) isdisposed between a first known direction and a second known direction.In one example, the first direction is equal to the 0° direction and thesecond direction is equal to the 30° direction, wherein the firstdirection extends from the vertex point 536 through a portion of a firstmicrophone 101A (e.g., geometric center of the microphone) and thesecond direction extends from the vertex point 536 in the 30° direction.

In one embodiment, to determine which of the quasi-directions theaudible source's direction is closest to, the direction determinationelement 463 first calculates ratios of various time delays (e.g., firstnon-zero delay/second non-zero delay) measured by the delay analysiselement 462 and then compares these calculated ratios with angular timedelay ratio data that is stored within the memory unit 122. The storedangular time delay ratio data will include previously calculated datathat is formed by calculating a ratio of the expected delays times thatthe microphones would see if the audible source was positioned at theedges of the bins that surround each of the quasi directions within adetermined region of the electronic device 102. Therefore, using theexample above, if the audible source is positioned at an angle of about35° relative to the electronic device 102, the direction determinationelement 463 will determine, based on a calculated ratio of the delaytime experienced by the microphone 101C to the delay time experienced bythe microphone 101B, that the calculated ratio is closer to a storedangular time delay ratio associated with the 30° quasi-direction thanany of the other stored angular time delay ratios associated with theother quasi-directions 0°, 60°, 90° or 120°. Alternately, the directiondetermination element 463 may determine that the calculated ratio iscloser to a stored angular time delay ratio associated with the 30°quasi-direction by determining that the calculated ratio falls within arange that is half the bin size on either side of the quasi-direction.In this example, the direction determination element 463 may compare thecalculated ratio with the stored angular time delay ratio associatedwith the directions 15°, 45°, 75° and 105°, and then determine that thecalculated ratio falls between the stored angular time delay ratios thatcoincide with 15° and 45°. Therefore, the audible source is most likelypositioned at the 30° quasi-direction.

In the somewhat rare case that the direction determination element 463finds that the calculated ratio exactly matches a stored angular timedelay ratio the controlling software program need not continue on withthe process of determining that the calculated ratio falls between thestored angular time delay ratios.

One will appreciate that the process of determining the direction of theaudible source is thus greatly simplified versus the mathematicallyintensive iterative process of determining the exact position of theaudible source using a more conventional analytically intensive andpower intensive approach. The greatly simplified statistical approach ofdetermining the source direction will also reduce the performancerequirements that the processor 118 needs to possess to perform thesetasks.

In some embodiments, the direction determination element 463 willdetermine a direction of a received audible signal by first determiningthe relative time delays experienced by each microphone, and thencomparing each of the relative time delays with a plurality of storedangular time delays. Each of the plurality of stored angular timedelays, which are stored within memory, can be associated with adirection that is oriented relative to the non-linear array ofmicrophones. Thus, for example, a stored angular time delay for eachmicrophone can be associated for each quasi-direction, such asquasi-directions 0°, 30°, 60°, 90° or 120°. However, it is believed thatthe use of the ratio of the expected delay times in certain geometricalarray configurations can be advantageous. For example, use of the ratiowhen the audible source may be positioned in 3-D space at an anglerelative to a plane that contains a planar array of microphones can beuseful due to the inherent comparison of the relative delays betweenmicrophones provided by the ratio versus other techniques that onlycompare delay times with the stored angular delay times.

However, since the accuracy of the time delay measurements determined bythe delay analysis element 462 is also limited by the number of samplesthat can be collected by the processor 118 within the actual time delayexperienced between microphones, some uncertainty in the determined timedelay values will exist. The accuracy of the time delay is thus limitedby the sampling frequency and spacing between microphones. The spacingof the microphones and the sampling frequency thus need to be largeenough to allow at least two samples to be taken within time that thereceipt of the audible signal is delayed without upsampling. One willnote that the process of upsampling can be a significant drain on theprocessor resources and also the electrical power required to performthis task. For example, if the processor is sampling at a frequency of48 kHz (e.g., 21 μs per sample) and the microphones are spaced 70 mmapart will allow 10 samples to be taken by the processor within thedelay time, while a microphone spacing of 14 mm would only allow theprocessor to take 2 samples within the delay time. The uncertainty inthe determined time delay values due to the often small number ofsamples and noise contained within the received audible signals cancause jitter between the determined source position states, which willaffect the ability of the direction determination element 463 todetermine and settle on one probable quasi-direction. Oscillatingbetween the determined source position states at a high rate may affectthe signal processing technique's ability to perform its desiredfunction.

In some embodiments of step 485, the controlling software programanalyzes the frequencies at which the various determined directions areselected by the direction determination element 463 to determine themost probable determined direction of the audible source. In oneexample, the controlling software program will compare the number oftimes various determined directions are selected by the directiondetermination element 463 over a period of time and then select thedirection that has the highest frequency over that period of time as thedetermined direction. Determining the most probable determined directioncan be performed in a rolling average type of process where eachdetermined direction within the rolling period can be taken as a “vote”that are summed to determine which direction gets the most votes overthe current rolling period. The frequency that each particulardetermined direction is determined may include the analysis of two ormore audio data samples that are sampled by the processor at the datasampling frequency (e.g., 48 kHz sampling frequency). This process candiminish the amount of jitter experienced from the output of thedirection determination element 463.

Referring to FIG. 4E and step 486, in an effort to minimize theuncertainty in the determined quasi-direction received from directiondetermination element 463, the coefficient generation element 464 isused to apply various weights to a number of the directions that areclose to the determined quasi-direction found during prior steps. FIG.5C graphically illustrates some of the coefficient values that may beused in our example above to weight the determination that an audiblesource, which is oriented at a 35° angle to the electronic device, has aquasi-direction that is 30°. In this example, the quasi-direction of 30°may receive a coefficient value of V₁ and the adjacent quasi-directionsof 0° and 60° may each receive a coefficient value of V₂ and V₃,respectively, which coincide with the probability that the actualdirection determined over a few audible signal data samples is alongeach of the quasi-directions. The coefficient values V₁, V₂ and V₃ canbe set at some value that is a percentage of full scale, and thus may,for example, have a value that includes real numbers between zero andone. The coefficient values may be determined based on how close thedetermined quasi-direction is to a stored angular time delay ratioand/or by forming a rolling average taken over a number of samples.

Next, at step 487, the direction selection element 465 then uses thedetermined coefficient values to determine the probable quasi-direction.The determined coefficient values are used to weight and thus damp thejitter experienced from the output of the direction determinationelement 463 as it determines and then refines the determinedquasi-directions every received audible signal data sample or couple ofaudible signal data samples. In some embodiments, the determinedcoefficient values for each of the probable directions are summed overthe sampling period or delay period, and the quasi-direction that hasthe highest sum total over the period is selected as the probablequasi-direction. In some configurations it may also be useful to giveall of the coefficients associated with non-likely directions a zero ornegative coefficient value to decrease the likelihood that thesedirections will be selected in this step.

At step 488, the direction delivery element 466 then delivers thedetected direction information, which contains the determined directionor quasi-direction, to the first signal processor 405 and the secondsignal processor 407. The detected direction information is thenreceived by the first signal processor 405 and the second signalprocessor 407, during steps 472 and 491, respectively, for further useor processing.

Second Signal Processor Example

Referring to FIGS. 4A and 4B, the second signal processor 407 isgenerally adapted to separate a desired audible signal from any unwantednoise that may be found within the higher end of the audible signalfrequency range by processing the audible signals received from thegeometrical array of microphones using a beamforming noise rejectiontechnique. FIG. 4F illustrates a method 490 of performing thebeamforming noise rejection technique, or second signal processingtechnique, using the second signal processor 407. The second signalprocessor 407 includes or uses portions of the controlling softwareprogram and various analog and digital hardware components to performthe desired processes described herein. In some embodiments, the secondsignal processor 407 includes elements that are formed within a digitalsignal processor (DSP) module. The beamforming processing techniqueperformed by the second signal processor 407 generally separates thenoise received from an off axis direction using an audible signal delayand summing technique using the audible inputs received from thegeometrical array of microphones. In some embodiments, the beamformingprocess is achieved by determining a delay for each microphone based onthe direction or quasi-direction determined by the direction detectionelement 430 and then summing the delayed audible signals received byeach of the microphones within the geometrical array of microphones sothat the audible signal received by the microphones from the audiblesource are all in phase and thus constructively add when combined.

In some embodiments, the second signal processor 407 uses the receiveddirection or quasi-direction information for a first period of time thatis longer than the time it takes the direction detection element 430 toupdate the direction or quasi-direction information (e.g., a secondtime). In this case, the rate at which the time delays are updatedduring the beamforming process is less than the rate at which thedirection detection element 430 is able to update the direction orquasi-direction information, which will reduce a significant amountcomputing power, battery power and time expended by the electronicdevice 102. Use of this process can be helpful to smooth the finalprocessed audible signal, which is generally not achieved if the ratethe direction is updated is too rapid.

FIG. 7A illustrates an example of a first beamforming pattern 701 thatis formed by the second signal processing technique using the pluralityof microphones that are a fixed distance apart (e.g., 70 mm) at a firstfrequency of about 100 Hz. One will note that a desired audible sourceis positioned at the −30° position relative to the orientation of thepolar graph. However, the beamforming technique at this low frequencyhas no ability to separate off-axis noise from the desired audiblesource, as shown by the circular shape of the beamforming pattern. FIG.7B illustrates an example of a second beamforming pattern 702 that isformed by the second signal processing technique at a second frequencyof about 1300 Hz using the same microphones used to form the firstbeamforming pattern 701. In this example, the second beam formingpattern 702, which is formed at a frequency of 1300 Hz, has a nearcardioid shape that is pointing in the direction of the desired audiblesource (e.g., −30°). FIG. 7C illustrates an example of a thirdbeamforming pattern 703 that is formed at a third frequency of about5000 Hz by the same microphones used to form the first beamformingpattern 701. In this example, the third beam forming pattern 703 hasalso become distorted due to the effect of frequency on the detectedsignal such that it contains eight lobes 703A-703H. The ability of thesecond signal processing technique to reject off-axis noise sources inall but three of the eight off-axis directions (e.g., lobes 703D-703F)at this frequency will be substantial, while the lobe 703A which isoriented in the audible sources direction will accept most if not all ofthe audible signal received from a desired audible source that ispositioned at the −30° position. As noted above, in some embodiments,the relative spacing between the various microphones may be adjusted inthe audible signal processing system 100 so that the frequency rangethat the second signal processing technique can operate within can bedesirably set. For example, the lower end of the frequency range, inwhich the second signal processing technique is able to adequatelyreject off-axis noise, may be selected such that it is between about1,000 Hz and about 4,000 Hz, while at the upper end of the frequencyrange, such as about 4000-8000 Hz, the second signal processingtechnique is able to form a dominant lobe (e.g., lobe 703A) in theaudible source's direction as similarly illustrated in FIG. 7C.

Referring back to FIGS. 4A and 4B, the second signal processor 407generally includes beamformer signal delaying elements 431A-431C, asignal combining element 432 and a signal filtering element 408,parametric equalizer 409, and a mix gain amplifier 410 disposed withinthe post processing element 451. Referring back to FIG. 4F, duringoperation, the second signal processor 407 first receives detecteddirection information from the direction detection element 430 (e.g.,step 491) that is based on audible signals received by the geometricalarray of microphones. The detected direction information then allows thebeamforming process to be aimed in the desired direction effectivelyreducing the amount of off-axis noise in the audible signals receivedafter passing through the various components within the directiondetection device 430, as described above. The beamforming process isgenerally performed by use of the audible signals received by all of themicrophones and the detected direction information received from thedirection detection element 430.

At steps 492-493, in some embodiments, the controlling software programperforms an analysis of the received detected direction informationreceived from the direction detection element 430 and then determinesbased on the detected direction or quasi-direction what the desireddelay needs to be for each of the audible signals received by each ofthe microphones based on specific directional time delay informationstored within the memory unit 122. The directional time delayinformation stored within the memory unit 122 may include a table of allof the possible directions or quasi-directions that the directiondetection element 430 will provide to the second signal processingtechnique and all of the time delay values that are associated with eachof the possible directions or quasi-directions for each of themicrophones. Thus, the table will contain a time delay value for eachmicrophone for each of the possible directions or quasi-directions. Inone example, as shown in FIG. 5A, the plurality of directions caninclude all directions, which at least have a component of the actualdirection, within a plane parallel to the top surface 106 (FIG. 1) fromall different angles (e.g., 0° to 360°). The time delay used by thesignal delaying elements 431A-431C is set by the known distance betweenthe microphones in the direction that the audible signal is received andthe known speed of sound.

Next, at step 494, the delayed audible signals are then combinedtogether by use of the signal combining element 432. Then, at step 495,the delayed audible signals are then optionally filtered by use of thesignal filtering element 408 within the post processing element 451. Thesignal filtering element may include a high pass filter that is able toremove frequencies that are lower than the useable range of the secondsignal processor 407. For example, high pass filter frequency may beconfigured to allow frequencies higher than about 1,000 Hz to pass. Theappropriately delayed and combined audible signals will constructivelyadd, and thus improve the signal-to-noise ratio of the audible source'saudible input versus the off-axis noise delivered from unwanted noisesources.

As noted above, after the first and second signal processing processeshave been completed the processed audible signal output from the firstsignal processor 405 may be further processed by use of a postprocessing elements 452 (step 477) and the processed audible signaloutput from the second signal processor 407 may be further processed byuse of a post processing elements 451 (step 495) before they arecombined together by the signal combining element 414 at step 498. Thepost processing elements 451, 452 may each include one or moreamplifiers that are able to adjust the signal levels of the processedaudible signals before they are combined.

FIGS. 8A-8C illustrate examples of various patterns that are formed atdifferent frequencies by use of a first signal processing technique anda second signal processing technique. The first and second signalprocessing techniques used to form the patterns found in FIGS. 8A-8Cemploy a geometrical microphone array that contains three microphonesthat are a fixed 70 mm distance apart. FIG. 8A illustrates an example ofa pattern 801 that is formed by the first signal processing techniqueusing a first-order cardioid based signal processing technique and apattern 802 that is formed by the second signal processing technique ata first frequency of about 100 Hz. One will note that the desiredaudible source is positioned at a 30° position relative to theorientation of the polar graph, such that the maximum amount of theaudible signal at this frequency is accepted by the first signalprocessing technique while the second signal processing technique isineffective at rejecting all audible signals at this frequency in alldirections. Therefore, the noise rejection provided by the first signalprocessing technique can be preferentially used to “clean up” theaudible signal at the low end frequencies.

FIG. 8B illustrates an example of a pattern 811 that is formed by thefirst signal processing technique using a first-order cardioid basedsignal processing technique and a pattern 812 that is formed by thesecond signal processing technique at a second frequency of about 1600Hz. One will note that by changing the frequency from 100 Hz to about1600 Hz the ability of the second signal processing technique to rejectoff-axis noise sources has improved, and as shown in FIG. 8B is similarto the cardioid pattern created by the first signal processing techniqueat this frequency. Since the noise rejection results received by both ofthe signal processing techniques are similar at 1600 Hz, a frequency ator near 1600 Hz can be selected and used by the controlling software asa transition point in the frequency range where the results receivedfrom first signal processing technique will be used at frequencies belowthe transition point and the results received from second signalprocessing technique will be used at frequencies above the transitionpoint. In some embodiments, it is desirable to select the high pass andlow pass filters used within the audible signal processing device 400 sothat the upper end of the frequency range of the first signal processingtechnique is set to a point that is at or at least close to thistransition frequency and the lower end of the frequency range of thesecond signal processing technique is set to a point that is at leastclose to this transition frequency.

FIG. 8C illustrates an example of a pattern 821 that is formed by thefirst signal processing technique using a first-order cardioid basedsignal processing technique and a pattern 822 that is formed by thesecond signal processing technique at a third frequency of about 6500Hz. One will note that the desired audible source is positioned at the30° position relative to the orientation of the polar graph, such thatthe maximum amount of the audible signal at this frequency is acceptedby the second signal processing technique (i.e., pattern 822) whilenearly all of audible signal at this frequency is rejected in audiblesource's directions by the first signal processing technique and theoff-axis side-lobe directions (e.g., angles −38°, −95°, 98° and 155°)are actually favored over the audible source's direction. Therefore, thenoise rejection provided by the second signal processing technique canbe preferentially used to “clean up” the audible signal at the high endfrequencies.

At step 499, the signal combining element 414 then provides theprocessed audible signal (e.g., desired audible signal) to a downstreamelement 415. As noted above, the downstream element 415 may include asoftware application or other electronic device that uses the processedaudible signal to perform some desired activity. The downstream element415 can be an electronic component that is in direct communication orwireless communication with the signal combining element 414, which isdisposed within the I/O device 116. In one configuration, the downstreamelement 415 can be an electronic component disposed within the audiblesignal processing system 100. In another configuration, the downstreamelement 415 can be an electronic component disposed within an electronicdevice that is external to the audible signal processing system 100.Examples of an external electronic device will include a wirelessspeaker, a video camera device, a keyboard, a smart phone, a speakerphone, a home automation device, or other useful electronic device thatis positioned to allow communication with one or more electroniccomponents found within the audible signal processing system 100.

One or more of the embodiments of the disclosure provided herein may beimplemented as a program product for use with a computer system. Theprogram(s) of the program product define functions of the embodiments(including the methods described herein) and can be contained on avariety of computer-readable storage media. Illustrativecomputer-readable storage media include, but are not limited to: (i)non-writable storage media (e.g., read-only memory devices within acomputer such as CD-ROM disks readable by a CD-ROM drive, flash memory,ROM chips or any type of solid-state non-volatile semiconductor memory)on which information is permanently stored; and (ii) writable storagemedia (e.g., floppy disks within a diskette drive or hard-disk drive orany type of solid-state random-access semiconductor memory) on whichalterable information is stored.

The invention has been described above with reference to specificembodiments. Persons skilled in the art, however, will understand thatvarious modifications and changes may be made thereto without departingfrom the broader spirit and scope of the invention as set forth in theappended claims. The foregoing description and drawings are,accordingly, to be regarded in an illustrative rather than a restrictivesense.

What is claimed is:
 1. An electronic device, comprising: a non-lineararray of microphones that comprises at least three microphones; adirection detection element that is configured to determine a directionof a first audible signal received by the microphones in the non-lineararray of microphones from an external audible source by use of acontrolling software program that is executed by a processor within theelectronic device; a first signal processor that is configured to removea first portion of an audible signal generated from a second audiblesource from a composite audible signal by use of the controllingsoftware program, wherein the first signal processor comprises: amicrophone selection element that is configured to select a firstmicrophone and a second microphone within the non-linear array ofmicrophones based on the determined direction received from thedirection detection element; a signal delay element configured toreceive the composite audible signal from the second microphone andproduce a delayed audible signal after a predetermined delay time haselapsed; a signal inverting element that is configured to receive thedelayed audible signal and produce an inverted audible signal that isbased on the delayed audible signal; and a first signal combiningelement that is configured to combine the inverted audible signal andthe composite audible signal received by the first microphone, andproduce a first combined audible signal; a second signal processor thatis configured to remove a second portion of the audible signal generatedfrom the second audible source from the composite audible signal by useof the controlling software program, wherein the second signal processorcomprises: a signal delay element that is configured to receive thecomposite audible signal from at least two of the microphones within thenon-linear array of microphones and produce a second delayed audiblesignal for each of the at least two of the microphones based on thedetermined direction received from the direction detection element; anda second signal combining element that is configured to combine each ofthe second delayed audible signals produced by the signal delayingelement and produce a second combined audible signal; and a third signalcombining element that is configured to combine the first combinedaudible signal and the second combined audible signal to produce a thirdcombined audible signal.
 2. The electronic device of claim 1, whereinthe third combined audible signal is transmitted to an electroniccomponent, and the electronic component is configured to perform anaudible signal processing activity based on the receipt of the thirdcombined audible signal.
 3. The electronic device of claim 2, whereinthe electronic component is disposed within a wireless speaker, videocamera device, keyboard, smart phone, speaker phone, or home automationdevice.
 4. The electronic device of claim 2, wherein the audible signalprocessing activity performed by the electronic component includes asoftware program that is adapted to perform a voice recognition, a voicedetection or a voice communication process.
 5. The electronic device ofclaim 2, wherein the determined direction is based on a comparison of atime delay experienced by each of the microphones when receiving thefirst audible signal.
 6. The electronic device of claim 1, wherein thefirst microphone comprises a virtual microphone that is formed byaveraging an audible signal input received from a third microphone and afourth microphone that are disposed within the non-linear array ofmicrophones.
 7. The electronic device of claim 6, wherein the audiblesignal input comprises the composite audible signal.
 8. The electronicdevice of claim 1, wherein the non-linear array of microphones, thefirst signal processor, the second signal processor and the third signalcombining element all receive power from a battery.
 9. The electronicdevice of claim 1, further comprising: a first filter configured tofilter the first combined audible signal before combining the firstcombined audible signal and the second combined audible signal to formthe third combined signal, wherein filtering the first combined signalcomprises removing all frequencies greater than a first frequency fromthe first combined audible signal; and a second filter configured tofilter the second combined audible signal before combining the firstcombined audible signal and the second combined audible signal to formthe third combined signal, wherein filtering the second combined signalcomprises removing all frequencies less than a second frequency from thesecond combined audible signal.
 10. The electronic device of claim 9,wherein the first frequency or the second frequency are between about1,000 and 4,000 Hertz (Hz).